Remove spammy GetRTPStatistics() log.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 2834 matching lines @@
 }
 
 int Channel::GetRTPStatistics(CallStatistics& stats) {
   // --- RtcpStatistics
 
   // The jitter statistics is updated for each received RTP packet and is
   // based on received packets.
   RtcpStatistics statistics;
   StreamStatistician* statistician =
       rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
-  if (!statistician ||
-      !statistician->GetStatistics(&statistics,
-                                   _rtpRtcpModule->RTCP() == RtcpMode::kOff)) {
-    _engineStatisticsPtr->SetLastError(
-        VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
-        "GetRTPStatistics() failed to read RTP statistics from the "
-        "RTP/RTCP module");
+  if (statistician) {
+    statistician->GetStatistics(&statistics,
+                                _rtpRtcpModule->RTCP() == RtcpMode::kOff);
   }
 
   stats.fractionLost = statistics.fraction_lost;
   stats.cumulativeLost = statistics.cumulative_lost;
   stats.extendedMax = statistics.extended_max_sequence_number;
   stats.jitterSamples = statistics.jitter;
 
   // --- RTT
   stats.rttMs = GetRTT(true);
 
@@ skipping 782 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc